Sync from SUSE:SLFO:Main webrtc-audio-processing-0 revision 50bed665ff886f41a6b8a88c0523b1c9
This commit is contained in:
commit
cc396a0dcc
23
.gitattributes
vendored
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23
.gitattributes
vendored
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## Default LFS
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*.7z filter=lfs diff=lfs merge=lfs -text
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*.bsp filter=lfs diff=lfs merge=lfs -text
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*.bz2 filter=lfs diff=lfs merge=lfs -text
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*.gem filter=lfs diff=lfs merge=lfs -text
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*.gz filter=lfs diff=lfs merge=lfs -text
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*.jar filter=lfs diff=lfs merge=lfs -text
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*.lz filter=lfs diff=lfs merge=lfs -text
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*.lzma filter=lfs diff=lfs merge=lfs -text
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*.obscpio filter=lfs diff=lfs merge=lfs -text
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*.oxt filter=lfs diff=lfs merge=lfs -text
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*.pdf filter=lfs diff=lfs merge=lfs -text
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*.png filter=lfs diff=lfs merge=lfs -text
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*.rpm filter=lfs diff=lfs merge=lfs -text
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*.tbz filter=lfs diff=lfs merge=lfs -text
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*.tbz2 filter=lfs diff=lfs merge=lfs -text
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*.tgz filter=lfs diff=lfs merge=lfs -text
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*.ttf filter=lfs diff=lfs merge=lfs -text
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||||||
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*.txz filter=lfs diff=lfs merge=lfs -text
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*.whl filter=lfs diff=lfs merge=lfs -text
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*.xz filter=lfs diff=lfs merge=lfs -text
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*.zip filter=lfs diff=lfs merge=lfs -text
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*.zst filter=lfs diff=lfs merge=lfs -text
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1
baselibs.conf
Normal file
1
baselibs.conf
Normal file
@ -0,0 +1 @@
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|||||||
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libwebrtc_audio_processing1
|
90
big_endian_support.patch
Normal file
90
big_endian_support.patch
Normal file
@ -0,0 +1,90 @@
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diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc
|
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|
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc.than 2016-05-24 08:28:45.749940095 -0400
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+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_file.cc 2016-05-24 08:50:30.361020010 -0400
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@@ -64,9 +64,6 @@ WavReader::~WavReader() {
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}
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||||||
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size_t WavReader::ReadSamples(size_t num_samples, int16_t* samples) {
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-#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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-#error "Need to convert samples to big-endian when reading from WAV file"
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|
-#endif
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// There could be metadata after the audio; ensure we don't read it.
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num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
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num_samples_remaining_);
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@@ -76,6 +73,12 @@ size_t WavReader::ReadSamples(size_t num
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RTC_CHECK(read == num_samples || feof(file_handle_));
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RTC_CHECK_LE(read, num_samples_remaining_);
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num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
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+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
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+ //convert to big-endian
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+ for(size_t idx = 0; idx < num_samples; idx++) {
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+ samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
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+ }
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+#endif
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return read;
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}
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||||||
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@@ -120,10 +123,17 @@ WavWriter::~WavWriter() {
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||||||
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void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
|
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#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
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-#error "Need to convert samples to little-endian when writing to WAV file"
|
||||||
|
-#endif
|
||||||
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+ int16_t * le_samples = new int16_t[num_samples];
|
||||||
|
+ for(size_t idx = 0; idx < num_samples; idx++) {
|
||||||
|
+ le_samples[idx] = (samples[idx]<<8) | (samples[idx]>>8);
|
||||||
|
+ }
|
||||||
|
+ const size_t written =
|
||||||
|
+ fwrite(le_samples, sizeof(*le_samples), num_samples, file_handle_);
|
||||||
|
+ delete []le_samples;
|
||||||
|
+#else
|
||||||
|
const size_t written =
|
||||||
|
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
|
||||||
|
+#endif
|
||||||
|
RTC_CHECK_EQ(num_samples, written);
|
||||||
|
num_samples_ += static_cast<uint32_t>(written);
|
||||||
|
RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
|
||||||
|
diff -up webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc
|
||||||
|
--- webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc.than 2016-05-24 08:50:52.591379263 -0400
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|
+++ webrtc-audio-processing-0.2/webrtc/common_audio/wav_header.cc 2016-05-24 08:52:08.552606848 -0400
|
||||||
|
@@ -129,7 +129,39 @@ static inline std::string ReadFourCC(uin
|
||||||
|
return std::string(reinterpret_cast<char*>(&x), 4);
|
||||||
|
}
|
||||||
|
#else
|
||||||
|
-#error "Write be-to-le conversion functions"
|
||||||
|
+static inline void WriteLE16(uint16_t* f, uint16_t x) {
|
||||||
|
+ *f = ((x << 8) & 0xff00) | ( ( x >> 8) & 0x00ff);
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+static inline void WriteLE32(uint32_t* f, uint32_t x) {
|
||||||
|
+ *f = ( (x & 0x000000ff) << 24 )
|
||||||
|
+ | ((x & 0x0000ff00) << 8)
|
||||||
|
+ | ((x & 0x00ff0000) >> 8)
|
||||||
|
+ | ((x & 0xff000000) >> 24 );
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
|
||||||
|
+ *f = (static_cast<uint32_t>(a) << 24 )
|
||||||
|
+ | (static_cast<uint32_t>(b) << 16)
|
||||||
|
+ | (static_cast<uint32_t>(c) << 8)
|
||||||
|
+ | (static_cast<uint32_t>(d) );
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+static inline uint16_t ReadLE16(uint16_t x) {
|
||||||
|
+ return (( x & 0x00ff) << 8 )| ((x & 0xff00)>>8);
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+static inline uint32_t ReadLE32(uint32_t x) {
|
||||||
|
+ return ( (x & 0x000000ff) << 24 )
|
||||||
|
+ | ( (x & 0x0000ff00) << 8 )
|
||||||
|
+ | ( (x & 0x00ff0000) >> 8)
|
||||||
|
+ | ( (x & 0xff000000) >> 24 );
|
||||||
|
+}
|
||||||
|
+
|
||||||
|
+static inline std::string ReadFourCC(uint32_t x) {
|
||||||
|
+ x = ReadLE32(x);
|
||||||
|
+ return std::string(reinterpret_cast<char*>(&x), 4);
|
||||||
|
+}
|
||||||
|
#endif
|
||||||
|
|
||||||
|
static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
|
24
big_endian_support_2.patch
Normal file
24
big_endian_support_2.patch
Normal file
@ -0,0 +1,24 @@
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|||||||
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diff -up webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef webrtc-audio-processing-0.2/webrtc/typedefs.h
|
||||||
|
--- webrtc-audio-processing-0.2/webrtc/typedefs.h.typedef 2016-05-12 09:08:53.885000410 -0500
|
||||||
|
+++ webrtc-audio-processing-0.2/webrtc/typedefs.h 2016-05-12 09:12:38.006851953 -0500
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||||||
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@@ -48,7 +48,19 @@
|
||||||
|
#define WEBRTC_ARCH_32_BITS
|
||||||
|
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
#else
|
||||||
|
-#error Please add support for your architecture in typedefs.h
|
||||||
|
+/* instead of failing, use typical unix defines... */
|
||||||
|
+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
||||||
|
+#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#else
|
||||||
|
+#error __BYTE_ORDER__ is not defined
|
||||||
|
+#endif
|
||||||
|
+#if defined(__LP64__)
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#else
|
||||||
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
|
+#endif
|
||||||
|
#endif
|
||||||
|
|
||||||
|
#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
|
BIN
webrtc-audio-processing-0.3.1.tar.xz
(Stored with Git LFS)
Normal file
BIN
webrtc-audio-processing-0.3.1.tar.xz
(Stored with Git LFS)
Normal file
Binary file not shown.
125
webrtc-audio-processing-0.changes
Normal file
125
webrtc-audio-processing-0.changes
Normal file
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|||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Sep 28 09:56:45 UTC 2023 - Antonio Larrosa <alarrosa@suse.com>
|
||||||
|
|
||||||
|
- Rename the 0.3.1 version of the package to
|
||||||
|
webrtc-audio-processing-0 so we can keep it around while all
|
||||||
|
applications are ported to version 1.x (like baresip and dino).
|
||||||
|
There's no need to rename the devel package since the new version
|
||||||
|
uses dashes instead of underscores in the package name.
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon Aug 17 15:30:03 UTC 2020 - Dirk Mueller <dmueller@suse.com>
|
||||||
|
|
||||||
|
- update to 0.3.1:
|
||||||
|
* doc: file invalid reference to pulseaudio mailing list
|
||||||
|
* various build system fixes
|
||||||
|
- spec-cleaner run
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Fri Aug 2 08:23:00 UTC 2019 - Martin Liška <mliska@suse.cz>
|
||||||
|
|
||||||
|
- Use FAT LTO objects in order to provide proper static library.
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de
|
||||||
|
|
||||||
|
- Add baselibs.conf for gstreamer-plugins-bad-32bit
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Remove webrtc-aarch64.patch, no longer needed
|
||||||
|
- Adapt the rest of webrtc- patches to new arch naming
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Remove unneeded explicit version dependency for automake
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Update to 0.3
|
||||||
|
* build: enforce linking with --no-undefined, add explicit -lpthread
|
||||||
|
* build: Make sure files with SSE2 code are compiled with -msse2
|
||||||
|
- Remove no-undefined.patch
|
||||||
|
- Remove webrtc-audio-processing-0.2-x86_msse2.patch
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Add no-undefined.patch patch
|
||||||
|
https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
|
||||||
|
- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
|
||||||
|
- Adapt big_endian_support.patch to new version
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
|
||||||
|
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
|
||||||
|
- Add big_endian_support.patch
|
||||||
|
https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
- New automake version dependency >= 1.5
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com
|
||||||
|
|
||||||
|
- Update to 0.2:
|
||||||
|
Contains API breaking changes.
|
||||||
|
|
||||||
|
Upstream changes include:
|
||||||
|
* Rewritten AGC and voice activity detection
|
||||||
|
* Intelligibility enhancer
|
||||||
|
* Extended AEC filter
|
||||||
|
* Beamformer
|
||||||
|
* Transient suppressor
|
||||||
|
* ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
|
||||||
|
|
||||||
|
API changes:
|
||||||
|
* We no longer include a top-level audio_processing.h. The webrtc tree format
|
||||||
|
is used, so use webrtc/modules/audio_processing/include/audio_processing.h
|
||||||
|
* The top-level module_common_types.h has also been moved to
|
||||||
|
webrtc/modules/interface/module_common_types.h
|
||||||
|
* C++11 support is now required while compiling client code
|
||||||
|
* AudioProcessing::Create() does not take any arguments any more
|
||||||
|
* AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
|
||||||
|
* Stream parameters are now configured via StreamConfig and ProcessingConfig
|
||||||
|
rather than set_sample_rate(), set_num_channels(), etc.
|
||||||
|
* AudioFrame field names have changed
|
||||||
|
* Use config API for newer audio processing options
|
||||||
|
* Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
|
||||||
|
when using the intelligibility enhancer
|
||||||
|
* GainControl::set_analog_level_limits() is broken. The AGC implementation
|
||||||
|
hard codes 0-255 as the volume range
|
||||||
|
|
||||||
|
Other notes:
|
||||||
|
* The new audio processing parameters are not all tested, and a few are not
|
||||||
|
enabled upstream (in Chromium) either
|
||||||
|
* The rewritten AGC appears to be less sensitive, and it might make sense to
|
||||||
|
initialise the capture volume to something reasonable (33% or 50%, for
|
||||||
|
example) to make sure there is sufficient energy in the stream to trigger
|
||||||
|
the AGC mechanism
|
||||||
|
- Adapted all 3 arch patches
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com
|
||||||
|
|
||||||
|
- Add patch webrtc-aarch64.patch from algraf to add aarch64 support
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Wed Dec 19 10:39:23 CET 2012 - ro@suse.de
|
||||||
|
|
||||||
|
- add s390 and s390x to known platforms
|
||||||
|
by adding webrtc-s390x.patch
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com
|
||||||
|
|
||||||
|
- add ppc64 to known platforms
|
||||||
|
|
||||||
|
-------------------------------------------------------------------
|
||||||
|
Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org
|
||||||
|
|
||||||
|
- initial version (0.1)
|
||||||
|
|
138
webrtc-audio-processing-0.spec
Normal file
138
webrtc-audio-processing-0.spec
Normal file
@ -0,0 +1,138 @@
|
|||||||
|
# vim: set sw=4 ts=4 et nu:
|
||||||
|
#
|
||||||
|
# spec file for package webrtc-audio-processing
|
||||||
|
#
|
||||||
|
# Copyright (c) 2020 SUSE LLC
|
||||||
|
# Copyright (c) 2012 Pascal Bleser <pascal.bleser@opensuse.org>
|
||||||
|
#
|
||||||
|
# All modifications and additions to the file contributed by third parties
|
||||||
|
# remain the property of their copyright owners, unless otherwise agreed
|
||||||
|
# upon. The license for this file, and modifications and additions to the
|
||||||
|
# file, is the same license as for the pristine package itself (unless the
|
||||||
|
# license for the pristine package is not an Open Source License, in which
|
||||||
|
# case the license is the MIT License). An "Open Source License" is a
|
||||||
|
# license that conforms to the Open Source Definition (Version 1.9)
|
||||||
|
# published by the Open Source Initiative.
|
||||||
|
|
||||||
|
# Please submit bugfixes or comments via https://bugs.opensuse.org/
|
||||||
|
#
|
||||||
|
|
||||||
|
|
||||||
|
%define soname 1
|
||||||
|
# Please submit bugfixes or comments via http://bugs.opensuse.org/
|
||||||
|
Name: webrtc-audio-processing-0
|
||||||
|
Version: 0.3.1
|
||||||
|
Release: 0
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
License: BSD-3-Clause
|
||||||
|
Group: System/Libraries
|
||||||
|
URL: https://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
|
||||||
|
Source: http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-%{version}.tar.xz
|
||||||
|
Source1: baselibs.conf
|
||||||
|
# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
Patch1: big_endian_support.patch
|
||||||
|
# PATCH-FIX-UPSTREAN big_endian_support.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
|
||||||
|
Patch2: big_endian_support_2.patch
|
||||||
|
# PATCH-FIX-OPENSUSE webrtc-(ppc64|s390x|aarch64).patch
|
||||||
|
Patch100: webrtc-ppc64.patch
|
||||||
|
Patch101: webrtc-s390x.patch
|
||||||
|
BuildRequires: autoconf
|
||||||
|
BuildRequires: automake
|
||||||
|
BuildRequires: gcc-c++
|
||||||
|
BuildRequires: glibc-devel
|
||||||
|
BuildRequires: libtool
|
||||||
|
BuildRequires: make
|
||||||
|
BuildRequires: pkgconfig
|
||||||
|
BuildRequires: xz
|
||||||
|
|
||||||
|
%description
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
This is a compatibility package which should only be used by applications
|
||||||
|
that haven't be updated yet to the newer 1.x version.
|
||||||
|
|
||||||
|
%package -n libwebrtc_audio_processing%{soname}
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: System/Libraries
|
||||||
|
|
||||||
|
%description -n libwebrtc_audio_processing%{soname}
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
This is a compatibility package which should only be used by applications
|
||||||
|
that haven't be updated yet to the newer 1.x version.
|
||||||
|
|
||||||
|
%package -n libwebrtc_audio_processing-devel
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc_audio_processing%{soname} = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc_audio_processing-devel
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
This is a compatibility package which should only be used by applications
|
||||||
|
that haven't be updated yet to the newer 1.x version.
|
||||||
|
|
||||||
|
%package -n libwebrtc_audio_processing-devel-static
|
||||||
|
Summary: Real-Time Communication Library for Web Browsers
|
||||||
|
Group: Development/Libraries/C and C++
|
||||||
|
Requires: libwebrtc_audio_processing-devel = %{version}
|
||||||
|
|
||||||
|
%description -n libwebrtc_audio_processing-devel-static
|
||||||
|
WebRTC is an open source project that enables web browsers with Real-Time
|
||||||
|
Communications (RTC) capabilities via simple Javascript APIs. The WebRTC
|
||||||
|
components have been optimized to best serve this purpose.
|
||||||
|
|
||||||
|
WebRTC implements the W3C's proposal for video conferencing on the web.
|
||||||
|
|
||||||
|
This is a compatibility package which should only be used by applications
|
||||||
|
that haven't be updated yet to the newer 1.x version.
|
||||||
|
|
||||||
|
%prep
|
||||||
|
%setup -q -T -c "%{name}-%{version}"
|
||||||
|
xz --decompress --stdout "%{SOURCE0}" | tar xf - --strip-components=1
|
||||||
|
sed -i 's/\r$//' AUTHORS
|
||||||
|
%patch1 -p1
|
||||||
|
%patch2 -p1
|
||||||
|
%patch100
|
||||||
|
%patch101
|
||||||
|
|
||||||
|
%build
|
||||||
|
%global _lto_cflags %{_lto_cflags} -ffat-lto-objects
|
||||||
|
%configure
|
||||||
|
%make_build
|
||||||
|
|
||||||
|
%install
|
||||||
|
%make_install
|
||||||
|
|
||||||
|
find %{buildroot} -type f -name "*.la" -delete -print
|
||||||
|
|
||||||
|
%post -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
||||||
|
%postun -n libwebrtc_audio_processing%{soname} -p /sbin/ldconfig
|
||||||
|
|
||||||
|
%files -n libwebrtc_audio_processing%{soname}
|
||||||
|
%license COPYING
|
||||||
|
%doc AUTHORS NEWS README.md UPDATING.md
|
||||||
|
%{_libdir}/libwebrtc_audio_processing.so.%{soname}
|
||||||
|
%{_libdir}/libwebrtc_audio_processing.so.%{soname}.*.*
|
||||||
|
|
||||||
|
%files -n libwebrtc_audio_processing-devel
|
||||||
|
%{_includedir}/webrtc_audio_processing
|
||||||
|
%{_libdir}/libwebrtc_audio_processing.so
|
||||||
|
%{_libdir}/pkgconfig/webrtc-audio-processing.pc
|
||||||
|
|
||||||
|
%files -n libwebrtc_audio_processing-devel-static
|
||||||
|
%{_libdir}/libwebrtc_audio_processing.a
|
||||||
|
|
||||||
|
%changelog
|
17
webrtc-ppc64.patch
Normal file
17
webrtc-ppc64.patch
Normal file
@ -0,0 +1,17 @@
|
|||||||
|
Index: webrtc/typedefs.h
|
||||||
|
===================================================================
|
||||||
|
--- webrtc/typedefs.h.org
|
||||||
|
+++ webrtc/typedefs.h
|
||||||
|
@@ -47,6 +47,12 @@
|
||||||
|
#elif defined(__pnacl__)
|
||||||
|
#define WEBRTC_ARCH_32_BITS
|
||||||
|
#define WEBRTC_ARCH_LITTLE_ENDIAN
|
||||||
|
+#elif defined(__powerpc64__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#elif defined(__powerpc__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
|
#else
|
||||||
|
/* instead of failing, use typical unix defines... */
|
||||||
|
#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
15
webrtc-s390x.patch
Normal file
15
webrtc-s390x.patch
Normal file
@ -0,0 +1,15 @@
|
|||||||
|
--- webrtc/typedefs.h
|
||||||
|
+++ webrtc/typedefs.h
|
||||||
|
@@ -53,6 +53,12 @@
|
||||||
|
#elif defined(__powerpc__)
|
||||||
|
#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
#define WEBRTC_ARCH_32_BITS
|
||||||
|
+#elif defined(__s390x__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_64_BITS
|
||||||
|
+#elif defined(__s390__)
|
||||||
|
+#define WEBRTC_ARCH_BIG_ENDIAN
|
||||||
|
+#define WEBRTC_ARCH_32_BITS
|
||||||
|
#else
|
||||||
|
/* instead of failing, use typical unix defines... */
|
||||||
|
#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
|
Loading…
Reference in New Issue
Block a user